search for: higazi

...outer, in the same network asterisk is running at, takes the call. but we can't hear / talk with each other. Ay ideas to get this thing solved?! My general section in sip.conf: [general] port=5060 bindaddr=0.0.0.0 localnet=192.168.1.0/255.255.255.0 externip=85.183.112.3 externhost=voipfax.higazi-it.com allowtransfer=yes qualify=yes nat=yes [2006] type=friend secret=frank host=dynamic context=nurintern nat=no [2007] type=friend secret=jochen host=192.168.1.2 context=nurintern nat=yes

.../srv/rails/demo/public/.htaccess: Invalid command ''RewriteEngine'', perhaps mis-spelled or defined by a module not included in the server configuration I compared every thing with the rails-wiki for fedora but I dont find the problem. please help me !! thank and best regards R Higazi -- Posted via http://www.ruby-forum.com/.

Hi people! I want to save the 1st letter from the ${EXTEN} variable. I don't want to trim it, I want to RESAVE it into a new variable. Let us assume the ${EXTEN} contains: 0698332977 then I'd love to get the 0 I would thank you for all advises. Tamer

Hi people! I am not getting really smart. I get the SVN Edition of asterisk GUI interface, compiled and love to get it to run, what won't work. What am I doing wrong?! svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0 make make checkconfig make install and If I open one of the URLs: http://localhost:8088/asterisk/static/config/cfgbasic.html

Hi people! Do you guys know where to find a STUN Server Howto?! Why?! We all know, to get Asterisk behind an NAT Router to run, is a bit tricky, and you might have to fire a lot of holes in your firewall. However, I would appreciate it very much if somebody could give me great links of how to set up a STUN Server. Tamer

Hi people! I am coding a special sollution for that I need to know if I can send AT commands in the extensions.conf, to one subscriber. Is there a way doing this through asterisk 1.6 ?! For sure anybody of you, would as why I want to do that. I want to speak to my endsystem directly with AT commands. For any advise, I would thank you kindly. Tamer

Hi! I have a Grandstream VoIP Device, at which a DECT base with 2 cordless phones are connected. If a call is placed and made through one cordless phone the other cordless phone appears as busy. What I want: 1. The Base station of the DECT cordless phones, is connected at 1 FXS Port of my Grandstream Telephone Adapter. 2. I want to place and receive as many calls at the same time through 1 SIP

Hi people! I am looking for a h.323 implementation guide for asterisk. I looked in the doc folder of the latest asterisk source distribution and I didn't fund anything acording to this subject. If you guys could give me any advise, I would thank you. Tamer

Hi! I have problems building asterisk 1.6.0.6. ./configure --prefix=/usr make gets me: enerating embedded module rules ... [CC] extconf.c -> extconf.o In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49, from extconf.c:45: /usr/local/include/integers.h:50:67: error: srtp_config.h: No such file or directory In file

Hi people! I need help according getting asterisk 1.6.0.6 installed. I posted to digium, but it seems to be that it is not an error, but either I am not getting smart what I have to do, to get it solved (configured and installed as well). ./configure make gets me this output: In file included from /usr/local/include/datatypes.h:50, from /usr/local/include/err.h:49,

Hi people! Asterisk PBX (version 1.6.5): I have Asterisk behind a NAT (192.168.1.2) SIP Phone: A client behind NAT (192.168.1.3) Softphone: One other client somewhere in the internet (also behind an NAT). they want to speak with each other, and if they do, there is no sound. if softphone in the internet is no more behind a NAT router, it can hear SIP Phone but SIP Phone is not able to hear

Hi people, I have a Cisco ATA 186 which understands only the SCCP protocoll, therefore I am a pure beginner and I hope that anybody of you could help me. How will I configure the ATA which has 2 analog ports? For any support I would kindly thank you Tamer

Hi! I am interisted to dial out from the console with chan_alsa. Can somebody of you help me according this problem?! I added user the asterisk to "pulse" and "pulse-access", and it didn't change anything. alsa applications are routed by default to pulse. cat /etc/asound.conf pcm.!default { type pulse } ctl.!default { type pulse } What might be the problem?!